[OpenSER-Users] Missing RTP stream

[Morten Isaksen]

Hi!

canreinvite is set to no and the OpenSER/mediaproxy is "hiding" the
clients IP-addresses from Asterisk, so I am pretty sure that this is
not the issue.

On 9/20/07, Norman Brandinger <norm at goes.com> wrote:
> Hi Morten,
>
> Admittedly, I haven't looked closely at your trace.  However, based on
> the description you gave, the first place to look is at the "canrevite"
> setting in Asterisk sip.conf.  You might want to try "canreinvite=no"
> after reading up on this particular setting.
>
> Regards,
> Norm
>
>
> Morten Isaksen wrote:
> > Hi!
> >
> > I have a strange problem with a missing RTP stream between OpenSER and
> > Asterisk. I am not sure if it is OpenSER og Asterisk related.
> >
> > I have this setup
> >
> > Phone A (172.17.96.17) --
> >                                       \      Openser    --    Asterisk
> >       --    PSTN
> >                                       /      (192.168.0.6)   (192.168.0.3)
> > Phone B (172.17.96.10) --        (172.17.64.1)
> >
> > I also have a Mediaproxy running on OpenSER and I force every call to
> > use the Mediaproxy.
> >
> > I call from Phone A or B to the PSTN works fine and from PSTN to Phone
> > A or B it also works.
> >
> > I have the dialplan logic on my Asterisk server so I want calls from
> > Phone A to Phone B to pass the Asterisk server. And this is were I
> > have the problem. When the call is established the RTP stream is
> > missing between Mediaproxy and Asterisk. I only have a RTP stream
> > between the phones and Mediaproxy. As far as I can see the SIP
> > signalling is correct.
> >
> > The SIP traces is listed below. Can you spot the problem in this?
> >
> > I will buy a beer (or 5) at OpenSER training in Rome to anyone who can
> > help me solve this problem.
> >
> > SIP trace between the phones and OpenSER:
> >


-- 
Morten Isaksen
http://www.misak.dk/blog/